1. Field of Invention
The present invention relates in general to mixed analog and digital signal processing and in particular, to analog to digital converters with integral sample rate conversion and systems and methods using the same.
2. Background of Invention
In many applications, converting data from its native analog form into the digital domain for processing, storage and transmission provides the best overall system performance. One well known example is audio processing where analog audio is digitized through analog to digital (A/D) conversion and then processed, for example filtered or compressed, and then stored on a digital storage medium such as a compact disk (CD) or digital video disk (DVD). On playback, the digital data is decompressed, as required, reconverted to analog through digital to analog (D/A) conversion, and finally presented to the end user as audible tones.
According to the Nyquist Theorem, so long as the analog waveform is sampled during A/D conversion at a sampling frequency at least twice as high as the highest frequency component, that waveform can be successively reconstructed during subsequent D/A conversion. In actual practice, oversampling A/D and D/A converters are typically used because of their relative ease in implementation. For example, in an 8xc3x97 oversampling converter operating on data with a base sampling rate of 44.1 kHz, the data are sampled at a rate of 352.8 kHz. At the higher sampling rate, operations such as anti-aliasing filtering are easier since a substantial amount of the noise power is translated to frequency bands well above the band of the signal of interest.
Sample rate conversion is an additional problem which must be addressed when processing digitized analog data. For example, professional digital audio is typically recorded with a sampling rate of 48 kHz while typical playback devices operate with a base sampling rate of 44.1 kHz. Sample rate conversion, and specifically down-conversion, is therefore required to ensure that the recorded audio properly plays back. There are several existing sample rate conversion techniques, including decimation for lowering the sampling rate and interpolation for increasing the sampling rate. Notwithstanding, these techniques are still subject to some significant disadvantages including the need for substantial silicon area for fabricating the requisite interpolation/decimation filters, as well as limitations on the ability to convert to fractional sampling rates.
According to one embodiment of the principles, an integrated analog to digital and sample rate converter is disclosed which includes sampling circuitry for receiving an analog signal and generating a stream of digital samples at a first rate. A leaky integrator filter then removes quantization noise from the stream of digital samples thus allowing resampling to be carried out at its output. A digital filter then filters the resampled stream of samples to generate an output stream of samples at a second rate.
The leaky integrator filter has the substantial advantage of providing a relatively narrow pass band, relatively high stop band attenuation, and requiring very little silicon area. Additionally, the leaky integrator filter allows the constraints on the following digital filter stage to be substantially relaxed. Overall, the number of filter stages required to perform both analog to digital conversion and sample rate conversion is reduced which in turn reduces the amount of silicon area required during fabrication.